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Hi,<br>
<br>
Asterisk either need to know when the remote caller ends his call,<br>
or it must detect the silence. <br>
<br>
Simplest solution is to activate silence detection, see
voicemail.conf.<br>
You may need to do some testing to get the proper
"silencethreshold" setting. <br>
<br>
Also search the archive, this is a often discussed issue...<br>
<a href="http://mharc.lists.openservices.ca/archives/html/asterisk-users/" eudora="autourl">http://mharc.lists.openservices.ca/archives/html/asterisk-users/</a><br>
<br>
/Stig<br>
<br>
<br>
At 17:25 2004-04-07 +0800, you wrote: <br>
<font face="arial" size=2><blockquote type=cite cite>Hi all,</font><br>
<font size=3> <br>
</font><font face="arial" size=2>In my Asterisk setup, incoming calls
through Cisco PSTN gateway to Asterisk extensions sounds work fine. All
calls can be terminated properly after hangup. However, when calls were
forwarded to voicemail, after recording & hangup the PSTN calls and
cisco FXO port remained connected unless cisco port was manually shut/no
shut. # key used to hang up the call did NOT help. Did anyone experience
the same problem??</font><br>
<font size=3> <br>
</font><font face="arial" size=2>------------------</font><br>
<font size=3> <br>
</font><font face="arial" size=2>sip*CLI></font><br>
<font size=2> -- Executing
Answer("SIP/-0811b4b8", "") in new stack<br>
-- Executing Wait("SIP/-0811b4b8",
"1") in new stack<br>
-- Executing VoiceMail("SIP/-0811b4b8",
"u6917") in new stack<br>
-- Playing 'voicemail/default/6917/unavail' (language
'en')<br>
-- Playing 'vm-intro' (language 'en')<br>
-- Playing 'beep' (language 'en')<br>
-- x=0, open writing:
/var/spool/asterisk/voicemail/default/6917/INBOX/msg0003 format: gsm,
0x81254f8<br>
-- x=1, open writing:
/var/spool/asterisk/voicemail/default/6917/INBOX/msg0003 format: wav49,
0x80fb178<br>
-- x=2, open writing:
/var/spool/asterisk/voicemail/default/6917/INBOX/msg0003 format: wav,
0x811af70<br>
-- Playing 'vm-msgsaved' (language 'en')<br>
-- Executing Hangup("SIP/-0811b4b8",
"") in new stack<br>
== Spawn extension (sip, 6917, 4) exited non-zero on
'SIP/-0811b4b8'<br>
sip*CLI></font><br>
<font size=3> <br>
</font><font face="arial" size=2>---------------------------<br>
<br>
cisco#sh voice call<br>
1/0/1<br>
vtsp level 0 state = S_CONNECTvpm level 1
state = FXOLS_CONNECT vpm level 0 state = S_UP<br>
<br>
--------------------------<br>
<br>
dial-peer voice 999 voip<br>
destination-pattern 8...<br>
session protocol sipv2<br>
session target ipv4:10.1.1.1:5065<br>
session transport udp<br>
codec g711ulaw<br>
no vad<br>
!<br>
------------------------------------<br>
exten => 6917,1,Answer<br>
exten => 6917,2,Wait(1)<br>
exten => 6917,3,VoiceMail(u${EXTEN})<br>
exten => 6917,4,Hangup<br>
Thanks.<br>
Ben<br>
</blockquote><br>
</font><br>
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